SIP TAPI

[dhodgson1980]

Good Evening,

I have searched the forum and can see that some people have managed to get SIP TAPI to work with 3CX. I have downloaded SIP TAPI and installed/configured it, when I make a call my SIP phone rings however when I pick up the phone the call disconnects. Below are the logs.

20:52:15.296 Call::Terminate [CM503008]: Call(7): Call is terminated
20:52:15.296 Call::Terminate [CM503008]: Call(7): Call is terminated
20:52:15.156 Call::RouteFailed [CM503014]: Call(7): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
20:52:15.109 CallCtrl:nSelectRouteReq [CM503013]: Call(7): No known route to target: [sip:[email protected]]
20:52:13.984 CallCtrl:nLegConnected [CM503007]: Call(7): Device joined: sip:[email protected]:6200
20:52:13.984 CallCtrl:nLegConnected [CM503007]: Call(7): Device joined: sip:[email protected]:5066
20:52:10.593 CallCtrl:nSelectRouteReq [CM503004]: Call(7): Calling: Shared:211@[Dev:sip:[email protected]:5066, Dev:sip:[email protected]:6200]
20:52:10.546 CallCtrl:nIncomingCall [CM503001]: Call(7): Incoming call from Ext.211 to [sip:[email protected]]

number blanked out for obvious reasons.

I am using free version of 3CX ver 5

If anyone knows how to configure SIP TAPI I would appreciate any help.

Thanks

 

[silentfun]

your log state that you have a problem with the outbound rule

set up a outboundrule with "Calls from extension(s) " = empty for "Calls to Numbers starting with (Prefix)"="0785" trough your provider (don´t forget to set "Strip Digits" to "0")in "Route X"

Andy

 

[dhodgson1980]

Hi,

Thanks that worked, however when the call transfers to the phone the line is silent, no ring tone. I also made a test call to my mobile and let the Mobile VM kick in but I could not hear it on the phone.

Any further advice on this?

Normal calls work fine.

 

[silentfun]

there are some cases for this audio things

1. is firewall (fastest check is disable firewall - restart 3cx PBX and then try - don´t forget enable FireWall again after that - only open all needed ports)

2. audi codec

3. you can troggle PBX Delivers Audio (on or off)

if you can add some of the log perhaps we can tell you more.

optional you can get a extension on our server for testing in conference.

Andy

 

[dhodgson1980]

Hi,

I cannot turn off my firewall, it is built in to my router but i did put the 3CX server into a DMZ and still no effect.

Not sure how to change codecs phones always seem to use G711u.

Here is a snippet from the logs when the call is made...

19:42:27.250 Call::Terminate [CM503008]: Call(23): Call is terminated
19:42:01.921 Line:rintEndpointInfo [CM505003]: Provider:[SIPgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.16.2:5060]
19:41:54.671 CallCtrl:nSelectRouteReq [CM503004]: Call(23): Calling: VoIPline:10000@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]
19:41:53.390 CallCtrl:nLegConnected [CM503007]: Call(23): Device joined: sip:[email protected]:6201
19:41:53.375 CallCtrl:nLegConnected [CM503007]: Call(23): Device joined: sip:[email protected]:5066
19:41:50.703 CallCtrl:nSelectRouteReq [CM503004]: Call(23): Calling: Shared:211@[Dev:sip:[email protected]:5066, Dev:sip:[email protected]:6201]
19:41:50.656 CallCtrl:nIncomingCall [CM503001]: Call(23): Incoming call from Ext.211 to [sip:[email protected]]

Hope this helps

 

[silentfun]

do you use v5 build 3790 ? if not try backup and then uninstall and install latest an restore or try to install over.

if not there was a problem with transfer and no audio.

Build version 5.0.3790 8 January 2008

New Features

Music on hold when transferring from Digital receptionist.
Ability to bypass STUN server resolution by removing stun server entries from general settings page.
By default, 3CX will use both auth ID and external line number to identify source of call from a voip provider.
By default, 3CX will use both LineID and Gateway host to indentify source of call from a PSTN gateway.
By default, port will be set to :5060 when comparing host/port fields in source identification rules.
Complete generation of Grandstream phones provisioning configuration without the need to use the GrandStream tool.
Added templates for the following gateways: Patton SN-4112 (2-port Analog), Patton SN-4552 (1-port BRI), Patton SN-4960/E1 (1-port PRI E1), Patton SN-4960/T1 (1-port PRI T1)
Fixed

Firewall checker releases ports after use.
Now it is possible to check multiple source identification rules, previously only the first one was checked.
Removed "Route calls for this Bridge during office hours to" table as there was no use for it.
Build version 5.0.3752 19 December 2007

Fixed: Multiple outbound calls over a single VoIP Provider account now works
Improved handling of recognition of local devices and external devices
Improved log messages - more complete information is now presented to help with creating source identification rules and inbound SIP Header field maps
Improved caching engine
Removal of OpenVPN components in preparation for new proxy + tunnelling protocol to ease NAT traversal.
Build version 5.0.3648 7 December 2007

Fixed which causes systems installed in a DMZ or on a Public IP to not work correctly
Fixed several issues VOIP providers
Improved licensing information display
Added a dialog to ask for FQDN of server, to allow for use of FQDN name of server in phone configuration
Improved feedback of firewall checker
Fix a bug where by rejected calls would work against license limit.

------------------------
Andy

 

[dhodgson1980]

Hi,

I am using the latest version.

I suppose having TAPI ability is not overly importent however it would be good to get it sorted.

I don't know enough about the system to diagnose the fault myself so I am greatful for any help.

Regards

 

[silentfun]

sorry that i could not help jet but not so easy on distance.

what clients you use and are they on the same subnet then your server ?

please post some of the server status log that is generated when starting up.

Andy

 

[weebnuts]

with Version 5, I would just use the 3cx VOIP client. It seems to work fine. I only had to use Sip Tapi in version 3 of 3cx because the client didn't support Tapi in outlook.

 

[dhodgson1980]

Excellent, I have installed the 3CX VOIP client and the provided TAPI works like a dream.

Well that is unless I set the client to Desk Phone Mode, the client nor the phone ring, it only seems to work if it is in Headset mode. Is this just a limitation of the free edition or do I need to set something up?

Thanks again for all your help

 

[dhodgson1980]

Excellent, I have installed the 3CX VOIP client and the provided TAPI works like a dream.

Well that is unless I set the client to Desk Phone Mode, the client nor the phone ring, it only seems to work if it is in Headset mode. Is this just a limitation of the free edition or do I need to set something up?

Thanks again for all your help

Can anyone tell me if there are restrictions on the free versions with using the VoIP client in desk phone mode. When I make a call neither the client or phone ring.

Using a Grandstream Budge Tone 100 if that helps.

 

[weebnuts]

not sure, but the client is still in Beta, so hopefully it will be fixed soon.